asterisk users - Jun 2008 - Weird one way Audio situation

Hi list,

I'm having trouble with calls placed to the PSTN (through a TDM card),
sometimes (a lot indeed) when I dial a number the callee party can't hear me
at all.

My setup is:

Asterisk 1.4.20.1
Zaptel 1.4.11
libpri 1.4.4
Wanpipe 3.2.4

I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP
Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel
2.4.16.60-0.23-smp

I'm using the ulaw audio codec.

There is no NAT between the Asterisk Server and the Phones (the phone and
the server are in the same network segment).

What can it be???

Thanks in advance for any help/comment...


-- 
Raul
Linux Counter #156439
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On Tue, Jun 10, 2008 at 1:40 PM, Ra?l G?mez C. <nachogomez at gmail.com>
wrote:
> Hi list,
>
> I'm having trouble with calls placed to the PSTN (through a TDM card),
> sometimes (a lot indeed) when I dial a number the callee party can't
hear me
> at all.
>
> My setup is:
>
> Asterisk 1.4.20.1
> Zaptel 1.4.11
> libpri 1.4.4
> Wanpipe 3.2.4
>
> I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000
IP
> Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel
> 2.4.16.60-0.23-smp
>
> I'm using the ulaw audio codec.
>
> There is no NAT between the Asterisk Server and the Phones (the phone and
> the server are in the same network segment).
>
> What can it be???
>
> Thanks in advance for any help/comment...
>
>
> --
> Raul
> Linux Counter #156439
Is your Asterisk box dual homed?  Firewalled?  Any output from the CLI
with verbose turned on, that might help?  Turn on SIP debugging as
well.

Thanks,
Steve T